2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
11 * This library is free software; you can redistribute it and/or
12 * modify it under the terms of the GNU Library General Public
13 * License as published by the Free Software Foundation; either
14 * version 2 of the License, or (at your option) any later version.
16 * This library is distributed in the hope that it will be useful,
17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19 * Library General Public License for more details.
21 * You should have received a copy of the GNU Library General Public
22 * License along with this library; if not, write to the
23 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
24 * Boston, MA 02110-1301, USA.
29 * SECTION:element-rtpjitterbuffer
31 * This element reorders and removes duplicate RTP packets as they are received
32 * from a network source.
34 * The element needs the clock-rate of the RTP payload in order to estimate the
35 * delay. This information is obtained either from the caps on the sink pad or,
36 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
37 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
39 * The rtpjitterbuffer will wait for missing packets up to a configurable time
40 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
41 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
42 * property is set, lost packets will result in a custom serialized downstream
43 * event of name GstRTPPacketLost. The lost packet events are usually used by a
44 * depayloader or other element to create concealment data or some other logic
45 * to gracefully handle the missing packets.
47 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
48 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
51 * The jitterbuffer can also be configured to send early retransmission events
52 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
53 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
54 * sends a custom upstream event named GstRTPRetransmissionRequest when the
55 * packet is considered late. The initial expected packet arrival time is
56 * calculated as follows:
58 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
59 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
60 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
61 * packets with different rtptime.
63 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
64 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
65 * previously scheduled timeout is overwritten.
67 * - If seqnum N arrived, all seqnum older than
68 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
69 * immediately. This is to request fast feedback for abonormally reorder
70 * packets before any of the previous timeouts is triggered.
72 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
73 * event. After the initial timeout expires and the retransmission event is
74 * sent, the timeout is scheduled for
75 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
76 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
77 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
78 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
79 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
80 * retransmission requests are sent and the regular logic is performed to
81 * schedule a lost packet as discussed above.
83 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
86 * This element will automatically be used inside rtpbin.
89 * <title>Example pipelines</title>
91 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
92 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
93 * inserted into the pipeline to smooth out network jitter and to reorder the
94 * out-of-order RTP packets.
105 #include <gst/rtp/gstrtpbuffer.h>
106 #include <gst/net/net.h>
108 #include "gstrtpjitterbuffer.h"
109 #include "rtpjitterbuffer.h"
110 #include "rtpstats.h"
112 #include <gst/glib-compat-private.h>
114 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
115 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
117 /* RTPJitterBuffer signals and args */
120 SIGNAL_REQUEST_PT_MAP,
128 #define DEFAULT_LATENCY_MS 200
129 #define DEFAULT_DROP_ON_LATENCY FALSE
130 #define DEFAULT_TS_OFFSET 0
131 #define DEFAULT_DO_LOST FALSE
132 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
133 #define DEFAULT_PERCENT 0
134 #define DEFAULT_DO_RETRANSMISSION FALSE
135 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
136 #define DEFAULT_RTX_DELAY -1
137 #define DEFAULT_RTX_MIN_DELAY 0
138 #define DEFAULT_RTX_DELAY_REORDER 3
139 #define DEFAULT_RTX_RETRY_TIMEOUT -1
140 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
141 #define DEFAULT_RTX_RETRY_PERIOD -1
142 #define DEFAULT_RTX_MAX_RETRIES -1
143 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
144 #define DEFAULT_MAX_DROPOUT_TIME 60000
145 #define DEFAULT_MAX_MISORDER_TIME 2000
146 #define DEFAULT_RFC7273_SYNC FALSE
148 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
149 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
155 PROP_DROP_ON_LATENCY,
160 PROP_DO_RETRANSMISSION,
161 PROP_RTX_NEXT_SEQNUM,
164 PROP_RTX_DELAY_REORDER,
165 PROP_RTX_RETRY_TIMEOUT,
166 PROP_RTX_MIN_RETRY_TIMEOUT,
167 PROP_RTX_RETRY_PERIOD,
168 PROP_RTX_MAX_RETRIES,
170 PROP_MAX_RTCP_RTP_TIME_DIFF,
171 PROP_MAX_DROPOUT_TIME,
172 PROP_MAX_MISORDER_TIME,
176 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
178 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
180 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
183 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
185 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
186 GST_DEBUG ("waiting timer"); \
187 (priv)->waiting_timer = TRUE; \
188 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
189 (priv)->waiting_timer = FALSE; \
190 GST_DEBUG ("waiting timer done"); \
192 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
193 if (G_UNLIKELY ((priv)->waiting_timer)) { \
194 GST_DEBUG ("signal timer"); \
195 g_cond_signal (&(priv)->jbuf_timer); \
199 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
200 GST_DEBUG ("waiting event"); \
201 (priv)->waiting_event = TRUE; \
202 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
203 (priv)->waiting_event = FALSE; \
204 GST_DEBUG ("waiting event done"); \
205 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
208 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
209 if (G_UNLIKELY ((priv)->waiting_event)) { \
210 GST_DEBUG ("signal event"); \
211 g_cond_signal (&(priv)->jbuf_event); \
215 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
216 GST_DEBUG ("waiting query"); \
217 (priv)->waiting_query = TRUE; \
218 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
219 (priv)->waiting_query = FALSE; \
220 GST_DEBUG ("waiting query done"); \
221 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
224 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
225 (priv)->last_query = res; \
226 if (G_UNLIKELY ((priv)->waiting_query)) { \
227 GST_DEBUG ("signal query"); \
228 g_cond_signal (&(priv)->jbuf_query); \
233 struct _GstRtpJitterBufferPrivate
235 GstPad *sinkpad, *srcpad;
238 RTPJitterBuffer *jbuf;
240 gboolean waiting_timer;
242 gboolean waiting_event;
244 gboolean waiting_query;
252 gboolean timer_running;
253 GThread *timer_thread;
258 gboolean drop_on_latency;
261 gboolean do_retransmission;
262 gboolean rtx_next_seqnum;
265 gint rtx_delay_reorder;
266 gint rtx_retry_timeout;
267 gint rtx_min_retry_timeout;
268 gint rtx_retry_period;
269 gint rtx_max_retries;
270 gint max_rtcp_rtp_time_diff;
271 guint32 max_dropout_time;
272 guint32 max_misorder_time;
274 /* the last seqnum we pushed out */
275 guint32 last_popped_seqnum;
276 /* the next expected seqnum we push */
278 /* seqnum-base, if known */
280 /* last output time */
281 GstClockTime last_out_time;
282 /* last valid input timestamp and rtptime pair */
283 GstClockTime ips_dts;
285 GstClockTime packet_spacing;
289 /* the next expected seqnum we receive */
290 GstClockTime last_in_dts;
291 guint32 next_in_seqnum;
295 /* start and stop ranges */
296 GstClockTime npt_start;
297 GstClockTime npt_stop;
298 guint64 ext_timestamp;
299 guint64 last_elapsed;
300 guint64 estimated_eos;
307 /* clock rate and rtp timestamp offset */
311 gint64 prev_ts_offset;
313 /* when we are shutting down */
314 GstFlowReturn srcresult;
320 GstClockTime timer_timeout;
321 guint16 timer_seqnum;
322 /* the latency of the upstream peer, we have to take this into account when
323 * synchronizing the buffers. */
324 GstClockTime peer_latency;
328 /* some accounting */
330 guint64 num_duplicates;
331 guint64 num_rtx_requests;
332 guint64 num_rtx_success;
333 guint64 num_rtx_failed;
336 RTPPacketRateCtx packet_rate_ctx;
339 GstClockTime last_dts;
340 guint64 last_rtptime;
341 GstClockTime avg_jitter;
358 GstClockTime timeout;
359 GstClockTime duration;
360 GstClockTime rtx_base;
361 GstClockTime rtx_delay;
362 GstClockTime rtx_retry;
363 GstClockTime rtx_last;
367 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
368 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
369 GstRtpJitterBufferPrivate))
371 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
372 GST_STATIC_PAD_TEMPLATE ("sink",
375 GST_STATIC_CAPS ("application/x-rtp"
376 /* "clock-rate = (int) [ 1, 2147483647 ], "
377 * "payload = (int) , "
378 * "encoding-name = (string) "
382 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
383 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
386 GST_STATIC_CAPS ("application/x-rtcp")
389 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
390 GST_STATIC_PAD_TEMPLATE ("src",
393 GST_STATIC_CAPS ("application/x-rtp"
394 /* "payload = (int) , "
395 * "clock-rate = (int) , "
396 * "encoding-name = (string) "
400 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
402 #define gst_rtp_jitter_buffer_parent_class parent_class
403 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
405 /* object overrides */
406 static void gst_rtp_jitter_buffer_set_property (GObject * object,
407 guint prop_id, const GValue * value, GParamSpec * pspec);
408 static void gst_rtp_jitter_buffer_get_property (GObject * object,
409 guint prop_id, GValue * value, GParamSpec * pspec);
410 static void gst_rtp_jitter_buffer_finalize (GObject * object);
412 /* element overrides */
413 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
414 * element, GstStateChange transition);
415 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
416 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
417 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
419 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
420 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
424 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
425 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
428 /* sinkpad overrides */
429 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
430 GstObject * parent, GstEvent * event);
431 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
432 GstObject * parent, GstBuffer * buffer);
434 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
435 GstObject * parent, GstEvent * event);
436 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
437 GstObject * parent, GstBuffer * buffer);
439 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
440 GstObject * parent, GstQuery * query);
442 /* srcpad overrides */
443 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
444 GstObject * parent, GstEvent * event);
445 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
446 GstObject * parent, GstPadMode mode, gboolean active);
447 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
448 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
449 GstObject * parent, GstQuery * query);
452 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
454 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
455 gboolean active, guint64 base_time);
456 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
458 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
459 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
461 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
463 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
467 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
469 GObjectClass *gobject_class;
470 GstElementClass *gstelement_class;
472 gobject_class = (GObjectClass *) klass;
473 gstelement_class = (GstElementClass *) klass;
475 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
477 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
479 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
480 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
483 * GstRtpJitterBuffer:latency:
485 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
486 * for at most this time.
488 g_object_class_install_property (gobject_class, PROP_LATENCY,
489 g_param_spec_uint ("latency", "Buffer latency in ms",
490 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
491 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
493 * GstRtpJitterBuffer:drop-on-latency:
495 * Drop oldest buffers when the queue is completely filled.
497 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
498 g_param_spec_boolean ("drop-on-latency",
499 "Drop buffers when maximum latency is reached",
500 "Tells the jitterbuffer to never exceed the given latency in size",
501 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
503 * GstRtpJitterBuffer:ts-offset:
505 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
506 * This is mainly used to ensure interstream synchronisation.
508 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
509 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
510 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
511 G_MAXINT64, DEFAULT_TS_OFFSET,
512 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
515 * GstRtpJitterBuffer:do-lost:
517 * Send out a GstRTPPacketLost event downstream when a packet is considered
520 g_object_class_install_property (gobject_class, PROP_DO_LOST,
521 g_param_spec_boolean ("do-lost", "Do Lost",
522 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
523 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
526 * GstRtpJitterBuffer:mode:
528 * Control the buffering and timestamping mode used by the jitterbuffer.
530 g_object_class_install_property (gobject_class, PROP_MODE,
531 g_param_spec_enum ("mode", "Mode",
532 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
533 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
535 * GstRtpJitterBuffer:percent:
537 * The percent of the jitterbuffer that is filled.
539 g_object_class_install_property (gobject_class, PROP_PERCENT,
540 g_param_spec_int ("percent", "percent",
541 "The buffer filled percent", 0, 100,
542 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
544 * GstRtpJitterBuffer:do-retransmission:
546 * Send out a GstRTPRetransmission event upstream when a packet is considered
547 * late and should be retransmitted.
551 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
552 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
553 "Send retransmission events upstream when a packet is late",
554 DEFAULT_DO_RETRANSMISSION,
555 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
558 * GstRtpJitterBuffer:rtx-next-seqnum
560 * Estimate when the next packet should arrive and schedule a retransmission
562 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
563 * for packet N+1. So it will be requested if it does not arrive at the expected time.
564 * The expected time is calculated using the dts of N and the packet spacing.
568 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
569 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
570 "Estimate when the next packet should arrive and schedule a "
571 "retransmission request for it.",
572 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
575 * GstRtpJitterBuffer:rtx-delay:
577 * When a packet did not arrive at the expected time, wait this extra amount
578 * of time before sending a retransmission event.
580 * When -1 is used, the max jitter will be used as extra delay.
584 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
585 g_param_spec_int ("rtx-delay", "RTX Delay",
586 "Extra time in ms to wait before sending retransmission "
587 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
588 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
591 * GstRtpJitterBuffer:rtx-min-delay:
593 * When a packet did not arrive at the expected time, wait at least this extra amount
594 * of time before sending a retransmission event.
598 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
599 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
600 "Minimum time in ms to wait before sending retransmission "
601 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
602 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
604 * GstRtpJitterBuffer:rtx-delay-reorder:
606 * Assume that a retransmission event should be sent when we see
607 * this much packet reordering.
609 * When -1 is used, the value will be estimated based on observed packet
614 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
615 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
616 "Sending retransmission event when this much reordering (-1 automatic)",
617 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
618 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
620 * GstRtpJitterBuffer::rtx-retry-timeout:
622 * When no packet has been received after sending a retransmission event
623 * for this time, retry sending a retransmission event.
625 * When -1 is used, the value will be estimated based on observed round
630 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
631 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
632 "Retry sending a transmission event after this timeout in "
633 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
634 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
636 * GstRtpJitterBuffer::rtx-min-retry-timeout:
638 * The minimum amount of time between retry timeouts. When
639 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
640 * minimum interval between retry timeouts.
642 * When -1 is used, the value will be estimated based on the
647 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
648 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
649 "Minimum timeout between sending a transmission event in "
650 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
651 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
653 * GstRtpJitterBuffer:rtx-retry-period:
655 * The amount of time to try to get a retransmission.
657 * When -1 is used, the value will be estimated based on the jitterbuffer
658 * latency and the observed round trip time.
662 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
663 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
664 "Try to get a retransmission for this many ms "
665 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
666 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
668 * GstRtpJitterBuffer:rtx-max-retries:
670 * The maximum number of retries to request a retransmission.
672 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
673 * When -1 is used, the number of retransmission request will not be limited.
677 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
678 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
679 "The maximum number of retries to request a retransmission. "
680 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
681 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
683 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
684 g_param_spec_uint ("max-dropout-time", "Max dropout time",
685 "The maximum time (milliseconds) of missing packets tolerated.",
686 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
687 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
689 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
690 g_param_spec_uint ("max-misorder-time", "Max misorder time",
691 "The maximum time (milliseconds) of misordered packets tolerated.",
692 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
693 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
695 * GstRtpJitterBuffer:stats:
697 * Various jitterbuffer statistics. This property returns a GstStructure
698 * with name application/x-rtp-jitterbuffer-stats with the following fields:
704 * <classname>"rtx-count"</classname>:
705 * the number of retransmissions requested.
711 * <classname>"rtx-success-count"</classname>:
712 * the number of successful retransmissions.
718 * <classname>"rtx-per-packet"</classname>:
719 * average number of RTX per packet.
725 * <classname>"rtx-rtt"</classname>:
726 * average round trip time per RTX.
733 g_object_class_install_property (gobject_class, PROP_STATS,
734 g_param_spec_boxed ("stats", "Statistics",
735 "Various statistics", GST_TYPE_STRUCTURE,
736 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
739 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
741 * The maximum amount of time in ms that the RTP time in the RTCP SRs
742 * is allowed to be ahead of the last RTP packet we received. Use
743 * -1 to disable ignoring of RTCP packets.
747 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
748 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
749 "Maximum amount of time in ms that the RTP time in RTCP SRs "
750 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
751 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
752 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
754 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
755 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
756 "Synchronize received streams to the RFC7273 clock "
757 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
758 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
761 * GstRtpJitterBuffer::request-pt-map:
762 * @buffer: the object which received the signal
765 * Request the payload type as #GstCaps for @pt.
767 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
768 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
769 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
770 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
771 GST_TYPE_CAPS, 1, G_TYPE_UINT);
773 * GstRtpJitterBuffer::handle-sync:
774 * @buffer: the object which received the signal
775 * @struct: a GstStructure containing sync values.
777 * Be notified of new sync values.
779 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
780 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
781 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
782 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
783 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
786 * GstRtpJitterBuffer::on-npt-stop:
787 * @buffer: the object which received the signal
789 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
790 * the npt-stop position.
792 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
793 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
794 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
795 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
796 G_TYPE_NONE, 0, G_TYPE_NONE);
799 * GstRtpJitterBuffer::clear-pt-map:
800 * @buffer: the object which received the signal
802 * Invalidate the clock-rate as obtained with the
803 * #GstRtpJitterBuffer::request-pt-map signal.
805 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
806 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
807 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
808 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
809 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
812 * GstRtpJitterBuffer::set-active:
813 * @buffer: the object which received the signal
815 * Start pushing out packets with the given base time. This signal is only
816 * useful in buffering mode.
818 * Returns: the time of the last pushed packet.
820 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
821 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
822 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
823 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
824 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
827 gstelement_class->change_state =
828 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
829 gstelement_class->request_new_pad =
830 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
831 gstelement_class->release_pad =
832 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
833 gstelement_class->provide_clock =
834 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
835 gstelement_class->set_clock =
836 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
838 gst_element_class_add_static_pad_template (gstelement_class,
839 &gst_rtp_jitter_buffer_src_template);
840 gst_element_class_add_static_pad_template (gstelement_class,
841 &gst_rtp_jitter_buffer_sink_template);
842 gst_element_class_add_static_pad_template (gstelement_class,
843 &gst_rtp_jitter_buffer_sink_rtcp_template);
845 gst_element_class_set_static_metadata (gstelement_class,
846 "RTP packet jitter-buffer", "Filter/Network/RTP",
847 "A buffer that deals with network jitter and other transmission faults",
848 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
849 "Wim Taymans <wim.taymans@gmail.com>");
851 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
852 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
854 GST_DEBUG_CATEGORY_INIT
855 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
859 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
861 GstRtpJitterBufferPrivate *priv;
863 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
864 jitterbuffer->priv = priv;
866 priv->latency_ms = DEFAULT_LATENCY_MS;
867 priv->latency_ns = priv->latency_ms * GST_MSECOND;
868 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
869 priv->do_lost = DEFAULT_DO_LOST;
870 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
871 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
872 priv->rtx_delay = DEFAULT_RTX_DELAY;
873 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
874 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
875 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
876 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
877 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
878 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
879 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
880 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
881 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
884 priv->last_rtptime = -1;
885 priv->avg_jitter = 0;
886 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
887 priv->jbuf = rtp_jitter_buffer_new ();
888 g_mutex_init (&priv->jbuf_lock);
889 g_cond_init (&priv->jbuf_timer);
890 g_cond_init (&priv->jbuf_event);
891 g_cond_init (&priv->jbuf_query);
892 g_queue_init (&priv->gap_packets);
894 /* reset skew detection initialy */
895 rtp_jitter_buffer_reset_skew (priv->jbuf);
896 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
897 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
901 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
904 gst_pad_set_activatemode_function (priv->srcpad,
905 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
906 gst_pad_set_query_function (priv->srcpad,
907 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
908 gst_pad_set_event_function (priv->srcpad,
909 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
912 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
915 gst_pad_set_chain_function (priv->sinkpad,
916 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
917 gst_pad_set_event_function (priv->sinkpad,
918 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
919 gst_pad_set_query_function (priv->sinkpad,
920 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
922 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
923 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
925 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
928 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
930 #define ITEM_TYPE_BUFFER 0
931 #define ITEM_TYPE_LOST 1
932 #define ITEM_TYPE_EVENT 2
933 #define ITEM_TYPE_QUERY 3
935 static RTPJitterBufferItem *
936 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
937 guint seqnum, guint count, guint rtptime)
939 RTPJitterBufferItem *item;
941 item = g_slice_new (RTPJitterBufferItem);
948 item->seqnum = seqnum;
950 item->rtptime = rtptime;
956 free_item (RTPJitterBufferItem * item)
958 g_return_if_fail (item != NULL);
960 if (item->data && item->type != ITEM_TYPE_QUERY)
961 gst_mini_object_unref (item->data);
962 g_slice_free (RTPJitterBufferItem, item);
966 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
968 GList **l = user_data;
970 if (item->data && item->type == ITEM_TYPE_EVENT
971 && GST_EVENT_IS_STICKY (item->data)) {
972 *l = g_list_prepend (*l, item->data);
973 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
974 gst_mini_object_unref (item->data);
976 g_slice_free (RTPJitterBufferItem, item);
980 gst_rtp_jitter_buffer_finalize (GObject * object)
982 GstRtpJitterBuffer *jitterbuffer;
983 GstRtpJitterBufferPrivate *priv;
985 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
986 priv = jitterbuffer->priv;
988 g_array_free (priv->timers, TRUE);
989 g_mutex_clear (&priv->jbuf_lock);
990 g_cond_clear (&priv->jbuf_timer);
991 g_cond_clear (&priv->jbuf_event);
992 g_cond_clear (&priv->jbuf_query);
994 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
995 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
996 g_queue_clear (&priv->gap_packets);
997 g_object_unref (priv->jbuf);
999 G_OBJECT_CLASS (parent_class)->finalize (object);
1002 static GstIterator *
1003 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1005 GstRtpJitterBuffer *jitterbuffer;
1006 GstPad *otherpad = NULL;
1007 GstIterator *it = NULL;
1008 GValue val = { 0, };
1010 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1012 if (pad == jitterbuffer->priv->sinkpad) {
1013 otherpad = jitterbuffer->priv->srcpad;
1014 } else if (pad == jitterbuffer->priv->srcpad) {
1015 otherpad = jitterbuffer->priv->sinkpad;
1016 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1017 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1021 g_value_init (&val, GST_TYPE_PAD);
1022 g_value_set_object (&val, otherpad);
1023 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1024 g_value_unset (&val);
1031 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1033 GstRtpJitterBufferPrivate *priv;
1035 priv = jitterbuffer->priv;
1037 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1040 gst_pad_new_from_static_template
1041 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1042 gst_pad_set_chain_function (priv->rtcpsinkpad,
1043 gst_rtp_jitter_buffer_chain_rtcp);
1044 gst_pad_set_event_function (priv->rtcpsinkpad,
1045 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1046 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1047 gst_rtp_jitter_buffer_iterate_internal_links);
1048 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1049 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1051 return priv->rtcpsinkpad;
1055 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1057 GstRtpJitterBufferPrivate *priv;
1059 priv = jitterbuffer->priv;
1061 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1063 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1065 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1066 priv->rtcpsinkpad = NULL;
1070 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1071 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1073 GstRtpJitterBuffer *jitterbuffer;
1074 GstElementClass *klass;
1076 GstRtpJitterBufferPrivate *priv;
1078 g_return_val_if_fail (templ != NULL, NULL);
1079 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1081 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1082 priv = jitterbuffer->priv;
1083 klass = GST_ELEMENT_GET_CLASS (element);
1085 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1087 /* figure out the template */
1088 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1089 if (priv->rtcpsinkpad != NULL)
1092 result = create_rtcp_sink (jitterbuffer);
1094 goto wrong_template;
1101 g_warning ("rtpjitterbuffer: this is not our template");
1106 g_warning ("rtpjitterbuffer: pad already requested");
1112 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1114 GstRtpJitterBuffer *jitterbuffer;
1115 GstRtpJitterBufferPrivate *priv;
1117 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1118 g_return_if_fail (GST_IS_PAD (pad));
1120 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1121 priv = jitterbuffer->priv;
1123 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1125 if (priv->rtcpsinkpad == pad) {
1126 remove_rtcp_sink (jitterbuffer);
1135 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1141 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1143 return gst_system_clock_obtain ();
1147 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1149 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1151 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1153 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1157 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1159 GstRtpJitterBufferPrivate *priv;
1161 priv = jitterbuffer->priv;
1163 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1166 priv->clock_rate = -1;
1167 /* do not clear current content, but refresh state for new arrival */
1168 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1169 rtp_jitter_buffer_reset_skew (priv->jbuf);
1174 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1177 GstRtpJitterBufferPrivate *priv;
1178 GstClockTime last_out;
1179 RTPJitterBufferItem *item;
1184 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1185 active, GST_TIME_ARGS (offset));
1187 if (active != priv->active) {
1188 /* add the amount of time spent in paused to the output offset. All
1189 * outgoing buffers will have this offset applied to their timestamps in
1190 * order to make them arrive in time in the sink. */
1191 priv->out_offset = offset;
1192 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1193 GST_TIME_ARGS (priv->out_offset));
1194 priv->active = active;
1195 JBUF_SIGNAL_EVENT (priv);
1198 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1200 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1201 /* head buffer timestamp and offset gives our output time */
1202 last_out = item->dts + priv->ts_offset;
1204 /* use last known time when the buffer is empty */
1205 last_out = priv->last_out_time;
1213 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1215 GstRtpJitterBuffer *jitterbuffer;
1216 GstRtpJitterBufferPrivate *priv;
1221 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1222 priv = jitterbuffer->priv;
1224 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1226 caps = gst_pad_peer_query_caps (other, filter);
1228 templ = gst_pad_get_pad_template_caps (pad);
1230 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1235 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1237 intersect = gst_caps_intersect (caps, templ);
1238 gst_caps_unref (caps);
1239 gst_caps_unref (templ);
1243 gst_object_unref (jitterbuffer);
1249 * Must be called with JBUF_LOCK held
1253 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1256 GstRtpJitterBufferPrivate *priv;
1257 GstStructure *caps_struct;
1260 const gchar *ts_refclk, *mediaclk;
1262 priv = jitterbuffer->priv;
1264 /* first parse the caps */
1265 caps_struct = gst_caps_get_structure (caps, 0);
1267 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1269 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1270 * measure the amount of data in the buffer */
1271 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1274 if (priv->clock_rate <= 0)
1277 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1279 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1281 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1282 * can use this to track the amount of time elapsed on the sender. */
1283 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1284 priv->clock_base = val;
1286 priv->clock_base = -1;
1288 priv->ext_timestamp = priv->clock_base;
1290 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1293 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1294 /* first expected seqnum, only update when we didn't have a previous base. */
1295 if (priv->next_in_seqnum == -1)
1296 priv->next_in_seqnum = val;
1297 if (priv->next_seqnum == -1) {
1298 priv->next_seqnum = val;
1299 JBUF_SIGNAL_EVENT (priv);
1301 priv->seqnum_base = val;
1303 priv->seqnum_base = -1;
1306 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1308 /* the start and stop times. The seqnum-base corresponds to the start time. We
1309 * will keep track of the seqnums on the output and when we reach the one
1310 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1311 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1312 priv->npt_start = tval;
1314 priv->npt_start = 0;
1316 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1317 priv->npt_stop = tval;
1319 priv->npt_stop = -1;
1321 GST_DEBUG_OBJECT (jitterbuffer,
1322 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1323 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1325 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1326 GstClock *clock = NULL;
1327 guint64 clock_offset = -1;
1329 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1332 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1333 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1334 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1336 const gchar *host, *portstr;
1340 host = ts_refclk + sizeof ("ntp=") - 1;
1341 if (host[0] == '[') {
1343 portstr = strchr (host, ']');
1344 if (portstr && portstr[1] == ':')
1345 portstr = portstr + 1;
1349 portstr = strrchr (host, ':');
1353 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1357 hostname = g_strndup (host, (portstr - host));
1359 hostname = g_strdup (host);
1361 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1364 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1365 const gchar *domainstr =
1366 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1369 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1372 clock = gst_ptp_clock_new (NULL, domain);
1374 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1377 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1378 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1380 if (!g_str_has_prefix (mediaclk, "direct=")
1381 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1382 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1383 if (strstr (mediaclk, "rate=") != NULL) {
1384 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1389 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1391 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1399 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1404 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1410 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1412 GstRtpJitterBufferPrivate *priv;
1414 priv = jitterbuffer->priv;
1417 /* mark ourselves as flushing */
1418 priv->srcresult = GST_FLOW_FLUSHING;
1419 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1420 /* this unblocks any waiting pops on the src pad task */
1421 JBUF_SIGNAL_EVENT (priv);
1422 JBUF_SIGNAL_QUERY (priv, FALSE);
1427 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1429 GstRtpJitterBufferPrivate *priv;
1431 priv = jitterbuffer->priv;
1434 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1435 /* Mark as non flushing */
1436 priv->srcresult = GST_FLOW_OK;
1437 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1438 priv->last_popped_seqnum = -1;
1439 priv->last_out_time = -1;
1440 priv->next_seqnum = -1;
1441 priv->seqnum_base = -1;
1442 priv->ips_rtptime = -1;
1443 priv->ips_dts = GST_CLOCK_TIME_NONE;
1444 priv->packet_spacing = 0;
1445 priv->next_in_seqnum = -1;
1446 priv->clock_rate = -1;
1449 priv->estimated_eos = -1;
1450 priv->last_elapsed = 0;
1451 priv->ext_timestamp = -1;
1452 priv->avg_jitter = 0;
1453 priv->last_dts = -1;
1454 priv->last_rtptime = -1;
1455 priv->last_in_dts = 0;
1456 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1457 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1458 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1459 rtp_jitter_buffer_reset_skew (priv->jbuf);
1460 remove_all_timers (jitterbuffer);
1461 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1462 g_queue_clear (&priv->gap_packets);
1467 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1468 GstPadMode mode, gboolean active)
1471 GstRtpJitterBuffer *jitterbuffer = NULL;
1473 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1476 case GST_PAD_MODE_PUSH:
1478 /* allow data processing */
1479 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1481 /* start pushing out buffers */
1482 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1483 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1484 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1486 /* make sure all data processing stops ASAP */
1487 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1489 /* NOTE this will hardlock if the state change is called from the src pad
1490 * task thread because we will _join() the thread. */
1491 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1492 result = gst_pad_stop_task (pad);
1502 static GstStateChangeReturn
1503 gst_rtp_jitter_buffer_change_state (GstElement * element,
1504 GstStateChange transition)
1506 GstRtpJitterBuffer *jitterbuffer;
1507 GstRtpJitterBufferPrivate *priv;
1508 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1510 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1511 priv = jitterbuffer->priv;
1513 switch (transition) {
1514 case GST_STATE_CHANGE_NULL_TO_READY:
1516 case GST_STATE_CHANGE_READY_TO_PAUSED:
1518 /* reset negotiated values */
1519 priv->clock_rate = -1;
1520 priv->clock_base = -1;
1521 priv->peer_latency = 0;
1523 /* block until we go to PLAYING */
1524 priv->blocked = TRUE;
1525 priv->timer_running = TRUE;
1526 priv->timer_thread =
1527 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1530 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1532 /* unblock to allow streaming in PLAYING */
1533 priv->blocked = FALSE;
1534 JBUF_SIGNAL_EVENT (priv);
1535 JBUF_SIGNAL_TIMER (priv);
1542 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1544 switch (transition) {
1545 case GST_STATE_CHANGE_READY_TO_PAUSED:
1546 /* we are a live element because we sync to the clock, which we can only
1547 * do in the PLAYING state */
1548 if (ret != GST_STATE_CHANGE_FAILURE)
1549 ret = GST_STATE_CHANGE_NO_PREROLL;
1551 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1553 /* block to stop streaming when PAUSED */
1554 priv->blocked = TRUE;
1555 unschedule_current_timer (jitterbuffer);
1557 if (ret != GST_STATE_CHANGE_FAILURE)
1558 ret = GST_STATE_CHANGE_NO_PREROLL;
1560 case GST_STATE_CHANGE_PAUSED_TO_READY:
1562 gst_buffer_replace (&priv->last_sr, NULL);
1563 priv->timer_running = FALSE;
1564 unschedule_current_timer (jitterbuffer);
1565 JBUF_SIGNAL_TIMER (priv);
1566 JBUF_SIGNAL_QUERY (priv, FALSE);
1568 g_thread_join (priv->timer_thread);
1569 priv->timer_thread = NULL;
1571 case GST_STATE_CHANGE_READY_TO_NULL:
1581 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1584 gboolean ret = TRUE;
1585 GstRtpJitterBuffer *jitterbuffer;
1586 GstRtpJitterBufferPrivate *priv;
1588 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1589 priv = jitterbuffer->priv;
1591 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1593 switch (GST_EVENT_TYPE (event)) {
1594 case GST_EVENT_LATENCY:
1596 GstClockTime latency;
1598 gst_event_parse_latency (event, &latency);
1600 GST_DEBUG_OBJECT (jitterbuffer,
1601 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1604 /* adjust the overall buffer delay to the total pipeline latency in
1605 * buffering mode because if downstream consumes too fast (because of
1606 * large latency or queues, we would start rebuffering again. */
1607 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1608 RTP_JITTER_BUFFER_MODE_BUFFER) {
1609 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1613 ret = gst_pad_push_event (priv->sinkpad, event);
1617 ret = gst_pad_push_event (priv->sinkpad, event);
1624 /* handles and stores the event in the jitterbuffer, must be called with
1627 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1629 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1630 RTPJitterBufferItem *item;
1633 switch (GST_EVENT_TYPE (event)) {
1634 case GST_EVENT_CAPS:
1638 gst_event_parse_caps (event, &caps);
1639 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1642 case GST_EVENT_SEGMENT:
1643 gst_event_copy_segment (event, &priv->segment);
1645 /* we need time for now */
1646 if (priv->segment.format != GST_FORMAT_TIME)
1647 goto newseg_wrong_format;
1649 GST_DEBUG_OBJECT (jitterbuffer,
1650 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1654 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1661 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1662 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1663 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
1665 JBUF_SIGNAL_EVENT (priv);
1670 newseg_wrong_format:
1672 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1673 gst_event_unref (event);
1679 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1682 gboolean ret = TRUE;
1683 GstRtpJitterBuffer *jitterbuffer;
1684 GstRtpJitterBufferPrivate *priv;
1686 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1687 priv = jitterbuffer->priv;
1689 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1691 switch (GST_EVENT_TYPE (event)) {
1692 case GST_EVENT_FLUSH_START:
1693 ret = gst_pad_push_event (priv->srcpad, event);
1694 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1695 /* wait for the loop to go into PAUSED */
1696 gst_pad_pause_task (priv->srcpad);
1698 case GST_EVENT_FLUSH_STOP:
1699 ret = gst_pad_push_event (priv->srcpad, event);
1701 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1702 GST_PAD_MODE_PUSH, TRUE);
1705 if (GST_EVENT_IS_SERIALIZED (event)) {
1706 /* serialized events go in the queue */
1708 if (priv->srcresult != GST_FLOW_OK) {
1709 /* Errors in sticky event pushing are no problem and ignored here
1710 * as they will cause more meaningful errors during data flow.
1711 * For EOS events, that are not followed by data flow, we still
1712 * return FALSE here though.
1714 if (!GST_EVENT_IS_STICKY (event) ||
1715 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1716 goto out_flow_error;
1718 /* refuse more events on EOS */
1721 ret = queue_event (jitterbuffer, event);
1724 /* non-serialized events are forwarded downstream immediately */
1725 ret = gst_pad_push_event (priv->srcpad, event);
1734 GST_DEBUG_OBJECT (jitterbuffer,
1735 "refusing event, we have a downstream flow error: %s",
1736 gst_flow_get_name (priv->srcresult));
1738 gst_event_unref (event);
1743 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1745 gst_event_unref (event);
1751 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1754 gboolean ret = TRUE;
1755 GstRtpJitterBuffer *jitterbuffer;
1757 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1759 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1761 switch (GST_EVENT_TYPE (event)) {
1762 case GST_EVENT_FLUSH_START:
1763 gst_event_unref (event);
1765 case GST_EVENT_FLUSH_STOP:
1766 gst_event_unref (event);
1769 ret = gst_pad_event_default (pad, parent, event);
1777 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1778 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1779 * GST_FLOW_FLUSHING when the element is shutting down. On success
1780 * GST_FLOW_OK is returned.
1782 static GstFlowReturn
1783 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1787 GValue args[2] = { {0}, {0} };
1791 g_value_init (&args[0], GST_TYPE_ELEMENT);
1792 g_value_set_object (&args[0], jitterbuffer);
1793 g_value_init (&args[1], G_TYPE_UINT);
1794 g_value_set_uint (&args[1], pt);
1796 g_value_init (&ret, GST_TYPE_CAPS);
1797 g_value_set_boxed (&ret, NULL);
1799 JBUF_UNLOCK (jitterbuffer->priv);
1800 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1802 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1804 g_value_unset (&args[0]);
1805 g_value_unset (&args[1]);
1806 caps = (GstCaps *) g_value_dup_boxed (&ret);
1807 g_value_unset (&ret);
1811 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1812 gst_caps_unref (caps);
1814 if (G_UNLIKELY (!res))
1822 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1823 return GST_FLOW_ERROR;
1827 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1828 return GST_FLOW_FLUSHING;
1832 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1833 return GST_FLOW_ERROR;
1837 /* call with jbuf lock held */
1839 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1841 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1842 GstMessage *message = NULL;
1847 /* Post a buffering message */
1848 if (priv->last_percent != percent) {
1849 priv->last_percent = percent;
1851 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1852 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1859 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1861 GstRtpJitterBufferPrivate *priv;
1863 priv = jitterbuffer->priv;
1865 if (timestamp == -1)
1868 /* apply the timestamp offset, this is used for inter stream sync */
1869 timestamp += priv->ts_offset;
1870 /* add the offset, this is used when buffering */
1871 timestamp += priv->out_offset;
1877 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1879 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1880 TimerData *timer = NULL;
1883 len = priv->timers->len;
1884 for (i = 0; i < len; i++) {
1885 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1886 if (test->seqnum == seqnum && test->type == type) {
1895 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1897 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1899 if (priv->clock_id) {
1900 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1901 gst_clock_id_unschedule (priv->clock_id);
1902 priv->clock_id = NULL;
1907 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1909 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1910 GstClockTime test_timeout;
1912 if ((test_timeout = timer->timeout) == -1)
1915 if (timer->type != TIMER_TYPE_EXPECTED) {
1916 /* add our latency and offset to get output times. */
1917 test_timeout = apply_offset (jitterbuffer, test_timeout);
1918 test_timeout += priv->latency_ns;
1920 return test_timeout;
1924 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1926 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1928 if (priv->clock_id) {
1929 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1931 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1932 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1934 if (timeout == -1 || timeout < priv->timer_timeout)
1935 unschedule_current_timer (jitterbuffer);
1940 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1941 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1942 GstClockTime duration)
1944 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1948 GST_DEBUG_OBJECT (jitterbuffer,
1949 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1950 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1951 GST_TIME_ARGS (delay));
1953 len = priv->timers->len;
1954 g_array_set_size (priv->timers, len + 1);
1955 timer = &g_array_index (priv->timers, TimerData, len);
1958 timer->seqnum = seqnum;
1960 timer->timeout = timeout + delay;
1961 timer->duration = duration;
1962 if (type == TIMER_TYPE_EXPECTED) {
1963 timer->rtx_base = timeout;
1964 timer->rtx_delay = delay;
1965 timer->rtx_retry = 0;
1967 timer->num_rtx_retry = 0;
1968 recalculate_timer (jitterbuffer, timer);
1969 JBUF_SIGNAL_TIMER (priv);
1975 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1976 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1978 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1979 gboolean seqchange, timechange;
1982 seqchange = timer->seqnum != seqnum;
1983 timechange = timer->timeout != timeout;
1985 if (!seqchange && !timechange)
1988 oldseq = timer->seqnum;
1990 GST_DEBUG_OBJECT (jitterbuffer,
1991 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1992 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1994 timer->timeout = timeout + delay;
1995 timer->seqnum = seqnum;
1997 timer->rtx_base = timeout;
1998 timer->rtx_delay = delay;
1999 timer->rtx_retry = 0;
2002 timer->num_rtx_retry = 0;
2004 if (priv->clock_id) {
2005 /* we changed the seqnum and there is a timer currently waiting with this
2006 * seqnum, unschedule it */
2007 if (seqchange && priv->timer_seqnum == oldseq)
2008 unschedule_current_timer (jitterbuffer);
2009 /* we changed the time, check if it is earlier than what we are waiting
2010 * for and unschedule if so */
2011 else if (timechange)
2012 recalculate_timer (jitterbuffer, timer);
2017 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2018 guint16 seqnum, GstClockTime timeout)
2022 /* find the seqnum timer */
2023 timer = find_timer (jitterbuffer, type, seqnum);
2024 if (timer == NULL) {
2025 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2027 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2033 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2035 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2038 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2039 unschedule_current_timer (jitterbuffer);
2042 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2043 g_array_remove_index_fast (priv->timers, idx);
2048 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2050 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2051 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2052 g_array_set_size (priv->timers, 0);
2053 unschedule_current_timer (jitterbuffer);
2056 /* get the extra delay to wait before sending RTX */
2058 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2062 if (priv->rtx_delay == -1) {
2063 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2064 delay = DEFAULT_AUTO_RTX_DELAY;
2066 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2067 * packet spacing is a good margin */
2068 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2071 delay = priv->rtx_delay * GST_MSECOND;
2073 if (priv->rtx_min_delay > 0)
2074 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2079 /* we just received a packet with seqnum and dts.
2081 * First check for old seqnum that we are still expecting. If the gap with the
2082 * current seqnum is too big, unschedule the timeouts.
2084 * If we have a valid packet spacing estimate we can set a timer for when we
2085 * should receive the next packet.
2086 * If we don't have a valid estimate, we remove any timer we might have
2087 * had for this packet.
2090 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2091 GstClockTime dts, gboolean do_next_seqnum)
2093 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2094 TimerData *timer = NULL;
2097 /* go through all timers and unschedule the ones with a large gap, also find
2098 * the timer for the seqnum */
2099 len = priv->timers->len;
2100 for (i = 0; i < len; i++) {
2101 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2104 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2106 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
2107 test->type, test->seqnum, seqnum, gap);
2110 GST_DEBUG ("found timer for current seqnum");
2111 /* the timer for the current seqnum */
2113 /* when no retransmission, we can stop now, we only need to find the
2114 * timer for the current seqnum */
2115 if (!priv->do_retransmission)
2117 } else if (gap > priv->rtx_delay_reorder) {
2118 /* max gap, we exceeded the max reorder distance and we don't expect the
2119 * missing packet to be this reordered */
2120 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2121 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2125 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2126 && priv->do_retransmission && priv->rtx_next_seqnum;
2128 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2129 if (timer->num_rtx_retry > 0) {
2130 GstClockTime rtx_last, delay;
2132 /* we scheduled a retry for this packet and now we have it */
2133 priv->num_rtx_success++;
2134 /* all the previous retry attempts failed */
2135 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2136 /* number of retries before receiving the packet */
2137 if (priv->avg_rtx_num == 0.0)
2138 priv->avg_rtx_num = timer->num_rtx_retry;
2140 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2141 /* calculate the delay between retransmission request and receiving this
2142 * packet, start with when we scheduled this timeout last */
2143 rtx_last = timer->rtx_last;
2144 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2145 /* we have a valid delay if this packet arrived after we scheduled the
2147 delay = dts - rtx_last;
2148 if (priv->avg_rtx_rtt == 0)
2149 priv->avg_rtx_rtt = delay;
2151 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2155 GST_LOG_OBJECT (jitterbuffer,
2156 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2157 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2158 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2159 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2160 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2161 GST_TIME_ARGS (priv->avg_rtx_rtt));
2163 /* don't try to estimate the next seqnum because this is a retransmitted
2164 * packet and it probably did not arrive with the expected packet
2166 do_next_seqnum = FALSE;
2170 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2171 GstClockTime expected, delay;
2173 /* calculate expected arrival time of the next seqnum */
2174 expected = dts + priv->packet_spacing;
2176 delay = get_rtx_delay (priv);
2178 /* and update/install timer for next seqnum */
2180 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2183 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2184 expected, delay, priv->packet_spacing);
2186 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2187 /* if we had a timer, remove it, we don't know when to expect the next
2189 remove_timer (jitterbuffer, timer);
2194 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2197 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2199 /* we need consecutive seqnums with a different
2200 * rtptime to estimate the packet spacing. */
2201 if (priv->ips_rtptime != rtptime) {
2202 /* rtptime changed, check dts diff */
2203 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2204 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2205 GstClockTime old_packet_spacing = priv->packet_spacing;
2207 /* Biased towards bigger packet spacings to prevent
2208 * too many unneeded retransmission requests for next
2209 * packets that just arrive a little later than we would
2211 if (old_packet_spacing > new_packet_spacing)
2212 priv->packet_spacing =
2213 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2214 else if (old_packet_spacing > 0)
2215 priv->packet_spacing =
2216 (3 * new_packet_spacing + old_packet_spacing) / 4;
2218 priv->packet_spacing = new_packet_spacing;
2220 GST_DEBUG_OBJECT (jitterbuffer,
2221 "new packet spacing %" GST_TIME_FORMAT
2222 " old packet spacing %" GST_TIME_FORMAT
2223 " combined to %" GST_TIME_FORMAT,
2224 GST_TIME_ARGS (new_packet_spacing),
2225 GST_TIME_ARGS (old_packet_spacing),
2226 GST_TIME_ARGS (priv->packet_spacing));
2228 priv->ips_rtptime = rtptime;
2229 priv->ips_dts = dts;
2234 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2235 guint16 seqnum, GstClockTime dts, gint gap)
2237 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2238 GstClockTime total_duration, duration, expected_dts;
2241 GST_DEBUG_OBJECT (jitterbuffer,
2242 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2243 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2245 if (dts == GST_CLOCK_TIME_NONE) {
2246 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2250 /* the total duration spanned by the missing packets */
2251 if (dts >= priv->last_in_dts)
2252 total_duration = dts - priv->last_in_dts;
2256 /* interpolate between the current time and the last time based on
2257 * number of packets we are missing, this is the estimated duration
2258 * for the missing packet based on equidistant packet spacing. */
2259 duration = total_duration / (gap + 1);
2261 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2262 GST_TIME_ARGS (duration));
2264 if (total_duration > priv->latency_ns) {
2265 GstClockTime gap_time;
2269 GstClockTime gap_dur = gap * duration;
2270 if (gap_dur > priv->latency_ns)
2271 gap_time = gap_dur - priv->latency_ns;
2274 lost_packets = gap_time / duration;
2276 gap_time = total_duration - priv->latency_ns;
2280 /* too many lost packets, some of the missing packets are already
2281 * too late and we can generate lost packet events for them. */
2282 GST_DEBUG_OBJECT (jitterbuffer,
2283 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2284 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2285 gap, expected, seqnum, GST_TIME_ARGS (total_duration),
2286 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2287 GST_TIME_ARGS (gap_time));
2289 /* this timer will fire immediately and the lost event will be pushed from
2290 * the timer thread */
2291 if (lost_packets > 0) {
2292 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2293 priv->last_in_dts + duration, 0, gap_time);
2294 expected += lost_packets;
2295 priv->last_in_dts += gap_time;
2299 expected_dts = priv->last_in_dts + duration;
2301 if (priv->do_retransmission) {
2304 type = TIMER_TYPE_EXPECTED;
2305 /* if we had a timer for the first missing packet, update it. */
2306 if ((timer = find_timer (jitterbuffer, type, expected))) {
2307 GstClockTime timeout = timer->timeout;
2309 timer->duration = duration;
2310 if (timeout > (expected_dts + timer->rtx_retry)) {
2311 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2312 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2316 expected_dts += duration;
2319 type = TIMER_TYPE_LOST;
2322 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2323 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2324 expected_dts += duration;
2330 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2334 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2335 GstRtpJitterBufferPrivate *priv;
2337 priv = jitterbuffer->priv;
2339 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2342 if (priv->last_dts != -1)
2343 dtsdiff = dts - priv->last_dts;
2347 if (priv->last_rtptime != -1)
2348 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2352 priv->last_dts = dts;
2353 priv->last_rtptime = rtptime;
2357 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2360 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2362 diff = ABS (dtsdiff - rtpdiffns);
2364 /* jitter is stored in nanoseconds */
2365 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2367 GST_LOG_OBJECT (jitterbuffer,
2368 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2369 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2370 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2371 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2378 GST_DEBUG_OBJECT (jitterbuffer,
2379 "no dts or no clock-rate, can't calculate jitter");
2385 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2387 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2388 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2391 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2392 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2393 gst_rtp_buffer_unmap (&rtp_a);
2395 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2396 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2397 gst_rtp_buffer_unmap (&rtp_b);
2399 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2403 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2404 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2407 GstRtpJitterBufferPrivate *priv;
2408 guint gap_packets_length;
2409 gboolean reset = FALSE;
2411 priv = jitterbuffer->priv;
2413 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2415 guint32 prev_gap_seq = -1;
2416 gboolean all_consecutive = TRUE;
2418 g_queue_insert_sorted (&priv->gap_packets, buffer,
2419 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2421 for (l = priv->gap_packets.head; l; l = l->next) {
2422 GstBuffer *gap_buffer = l->data;
2423 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2426 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2428 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2430 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2431 if (prev_gap_seq == -1)
2432 prev_gap_seq = gap_seq;
2433 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2434 all_consecutive = FALSE;
2436 prev_gap_seq = gap_seq;
2438 gst_rtp_buffer_unmap (&gap_rtp);
2439 if (!all_consecutive)
2443 if (all_consecutive && gap_packets_length > 3) {
2444 GST_DEBUG_OBJECT (jitterbuffer,
2445 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2446 (future ? "new" : "old"), gap,
2447 (future ? max_dropout : -max_misorder));
2449 } else if (!all_consecutive) {
2450 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2451 g_queue_clear (&priv->gap_packets);
2452 GST_DEBUG_OBJECT (jitterbuffer,
2453 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2454 (future ? "new" : "old"), gap,
2455 (future ? max_dropout : -max_misorder));
2458 GST_DEBUG_OBJECT (jitterbuffer,
2459 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2460 (future ? "new" : "old"), gap,
2461 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2465 GST_DEBUG_OBJECT (jitterbuffer,
2466 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2467 gap, -max_misorder);
2468 g_queue_push_tail (&priv->gap_packets, buffer);
2476 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2478 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2479 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2482 GstClockTime base_time =
2483 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2484 GstClockTime clock_time = gst_clock_get_time (clock);
2486 if (clock_time > base_time)
2487 running_time = clock_time - base_time;
2491 gst_object_unref (clock);
2494 return running_time;
2497 static GstFlowReturn
2498 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2501 GstRtpJitterBuffer *jitterbuffer;
2502 GstRtpJitterBufferPrivate *priv;
2504 guint32 expected, rtptime;
2505 GstFlowReturn ret = GST_FLOW_OK;
2506 GstClockTime dts, pts;
2511 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2512 gboolean do_next_seqnum = FALSE;
2513 RTPJitterBufferItem *item;
2514 GstMessage *msg = NULL;
2515 gboolean estimated_dts = FALSE;
2516 guint32 packet_rate, max_dropout, max_misorder;
2518 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2520 priv = jitterbuffer->priv;
2522 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2523 goto invalid_buffer;
2525 pt = gst_rtp_buffer_get_payload_type (&rtp);
2526 seqnum = gst_rtp_buffer_get_seq (&rtp);
2527 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2528 gst_rtp_buffer_unmap (&rtp);
2530 /* make sure we have PTS and DTS set */
2531 pts = GST_BUFFER_PTS (buffer);
2532 dts = GST_BUFFER_DTS (buffer);
2539 /* If we have no DTS here, i.e. no capture time, get one from the
2540 * clock now to have something to calculate with in the future. */
2541 dts = get_current_running_time (jitterbuffer);
2544 /* Remember that we estimated the DTS if we are running already
2545 * and this is not our first packet (or first packet after a reset).
2546 * If it's the first packet, we somehow must generate a timestamp for
2547 * everything, otherwise we can't calculate any times
2549 estimated_dts = (priv->next_in_seqnum != -1);
2551 /* take the DTS of the buffer. This is the time when the packet was
2552 * received and is used to calculate jitter and clock skew. We will adjust
2553 * this DTS with the smoothed value after processing it in the
2554 * jitterbuffer and assign it as the PTS. */
2555 /* bring to running time */
2556 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2559 GST_DEBUG_OBJECT (jitterbuffer,
2560 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2561 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2563 JBUF_LOCK_CHECK (priv, out_flushing);
2565 if (G_UNLIKELY (priv->last_pt != pt)) {
2568 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2572 /* reset clock-rate so that we get a new one */
2573 priv->clock_rate = -1;
2575 /* Try to get the clock-rate from the caps first if we can. If there are no
2576 * caps we must fire the signal to get the clock-rate. */
2577 if ((caps = gst_pad_get_current_caps (pad))) {
2578 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2579 gst_caps_unref (caps);
2583 if (G_UNLIKELY (priv->clock_rate == -1)) {
2584 /* no clock rate given on the caps, try to get one with the signal */
2585 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2586 pt) == GST_FLOW_FLUSHING)
2589 if (G_UNLIKELY (priv->clock_rate == -1))
2593 /* don't accept more data on EOS */
2594 if (G_UNLIKELY (priv->eos))
2597 calculate_jitter (jitterbuffer, dts, rtptime);
2599 if (priv->seqnum_base != -1) {
2602 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2605 GST_DEBUG_OBJECT (jitterbuffer,
2606 "packet seqnum #%d before seqnum-base #%d", seqnum,
2608 gst_buffer_unref (buffer);
2611 } else if (gap > 16384) {
2612 /* From now on don't compare against the seqnum base anymore as
2613 * at some point in the future we will wrap around and also that
2614 * much reordering is very unlikely */
2615 priv->seqnum_base = -1;
2619 expected = priv->next_in_seqnum;
2622 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2624 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2625 priv->max_dropout_time);
2627 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2628 priv->max_misorder_time);
2629 GST_TRACE_OBJECT (jitterbuffer,
2630 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2631 max_dropout, max_misorder);
2633 /* now check against our expected seqnum */
2634 if (G_LIKELY (expected != -1)) {
2637 /* now calculate gap */
2638 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2640 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2641 expected, seqnum, gap);
2643 if (G_LIKELY (gap == 0)) {
2644 /* packet is expected */
2645 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2646 do_next_seqnum = TRUE;
2648 gboolean reset = FALSE;
2651 /* we received an old packet */
2652 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2654 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2655 gap, max_dropout, max_misorder);
2658 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2661 /* new packet, we are missing some packets */
2662 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2663 /* If we have timers for more than RTP_MAX_DROPOUT packets
2664 * pending this means that we have a huge gap overall. We can
2665 * reset the jitterbuffer at this point because there's
2666 * just too much data missing to be able to do anything
2667 * sensible with the past data. Just try again from the
2669 GST_WARNING_OBJECT (jitterbuffer,
2670 "%d pending timers > %d - resetting", priv->timers->len,
2673 gst_buffer_unref (buffer);
2675 } else if (G_UNLIKELY (gap >= max_dropout)) {
2677 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2678 gap, max_dropout, max_misorder);
2681 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2682 /* fill in the gap with EXPECTED timers */
2683 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2685 do_next_seqnum = TRUE;
2688 if (G_UNLIKELY (reset)) {
2689 GList *events = NULL, *l;
2692 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2693 rtp_jitter_buffer_flush (priv->jbuf,
2694 (GFunc) free_item_and_retain_events, &events);
2695 rtp_jitter_buffer_reset_skew (priv->jbuf);
2696 remove_all_timers (jitterbuffer);
2697 priv->discont = TRUE;
2698 priv->last_popped_seqnum = -1;
2700 if (priv->gap_packets.head) {
2701 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2702 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2704 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2705 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2706 gst_rtp_buffer_unmap (&gap_rtp);
2708 priv->next_seqnum = seqnum;
2711 priv->last_in_dts = -1;
2712 priv->next_in_seqnum = -1;
2714 /* Insert all sticky events again in order, otherwise we would
2715 * potentially loose STREAM_START, CAPS or SEGMENT events
2717 events = g_list_reverse (events);
2718 for (l = events; l; l = l->next) {
2719 RTPJitterBufferItem *item;
2721 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2722 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
2724 g_list_free (events);
2726 JBUF_SIGNAL_EVENT (priv);
2728 /* reset spacing estimation when gap */
2729 priv->ips_rtptime = -1;
2730 priv->ips_dts = GST_CLOCK_TIME_NONE;
2732 buffers = g_list_copy (priv->gap_packets.head);
2733 g_queue_clear (&priv->gap_packets);
2735 priv->ips_rtptime = -1;
2736 priv->ips_dts = GST_CLOCK_TIME_NONE;
2737 JBUF_UNLOCK (jitterbuffer->priv);
2739 for (l = buffers; l; l = l->next) {
2740 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2742 if (ret != GST_FLOW_OK)
2745 for (; l; l = l->next)
2746 gst_buffer_unref (l->data);
2747 g_list_free (buffers);
2751 /* reset spacing estimation when gap */
2752 priv->ips_rtptime = -1;
2753 priv->ips_dts = GST_CLOCK_TIME_NONE;
2756 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2758 /* we don't know what the next_in_seqnum should be, wait for the last
2759 * possible moment to push this buffer, maybe we get an earlier seqnum
2761 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2762 do_next_seqnum = TRUE;
2763 /* take rtptime and dts to calculate packet spacing */
2764 priv->ips_rtptime = rtptime;
2765 priv->ips_dts = dts;
2768 /* We had no huge gap, let's drop all the gap packets */
2769 if (buffer != NULL) {
2770 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2771 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2772 g_queue_clear (&priv->gap_packets);
2774 GST_DEBUG_OBJECT (jitterbuffer,
2775 "Had big gap, waiting for more consecutive packets");
2776 JBUF_UNLOCK (jitterbuffer->priv);
2780 if (do_next_seqnum) {
2781 priv->last_in_dts = dts;
2782 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2785 /* let's check if this buffer is too late, we can only accept packets with
2786 * bigger seqnum than the one we last pushed. */
2787 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2790 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2792 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2793 if (G_UNLIKELY (gap <= 0))
2797 /* let's drop oldest packet if the queue is already full and drop-on-latency
2798 * is set. We can only do this when there actually is a latency. When no
2799 * latency is set, we just pump it in the queue and let the other end push it
2800 * out as fast as possible. */
2801 if (priv->latency_ms && priv->drop_on_latency) {
2803 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2805 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2806 RTPJitterBufferItem *old_item;
2808 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2810 if (IS_DROPABLE (old_item)) {
2811 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2812 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2814 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2815 free_item (old_item);
2817 /* we might have removed some head buffers, signal the pushing thread to
2818 * see if it can push now */
2819 JBUF_SIGNAL_EVENT (priv);
2823 /* If we estimated the DTS, don't consider it in the clock skew calculations
2824 * later. The code above always sets dts to pts or the other way around if
2825 * any of those is valid in the buffer, so we know that if we estimated the
2826 * dts that both are unknown */
2829 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2830 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2832 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2834 /* now insert the packet into the queue in sorted order. This function returns
2835 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2836 * have a duplicate. */
2837 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2839 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)))))
2843 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2845 /* we had an unhandled SR, handle it now */
2847 do_handle_sync (jitterbuffer);
2849 if (G_UNLIKELY (head)) {
2850 /* signal addition of new buffer when the _loop is waiting. */
2851 if (G_LIKELY (priv->active))
2852 JBUF_SIGNAL_EVENT (priv);
2854 /* let's unschedule and unblock any waiting buffers. We only want to do this
2855 * when the head buffer changed */
2856 if (G_UNLIKELY (priv->clock_id)) {
2857 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2858 unschedule_current_timer (jitterbuffer);
2862 GST_DEBUG_OBJECT (jitterbuffer,
2863 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2864 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2866 msg = check_buffering_percent (jitterbuffer, percent);
2872 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2879 /* this is not fatal but should be filtered earlier */
2880 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2881 ("Received invalid RTP payload, dropping"));
2882 gst_buffer_unref (buffer);
2887 GST_WARNING_OBJECT (jitterbuffer,
2888 "No clock-rate in caps!, dropping buffer");
2889 gst_buffer_unref (buffer);
2894 ret = priv->srcresult;
2895 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2896 gst_buffer_unref (buffer);
2902 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2903 gst_buffer_unref (buffer);
2908 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2909 " popped, dropping", seqnum, priv->last_popped_seqnum);
2911 gst_buffer_unref (buffer);
2916 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2918 priv->num_duplicates++;
2925 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2927 guint64 ext_time, elapsed;
2929 GstRtpJitterBufferPrivate *priv;
2931 priv = jitterbuffer->priv;
2932 rtp_time = item->rtptime;
2934 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2935 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2937 ext_time = priv->ext_timestamp;
2938 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2939 if (ext_time < priv->ext_timestamp) {
2940 ext_time = priv->ext_timestamp;
2942 priv->ext_timestamp = ext_time;
2945 if (ext_time > priv->clock_base)
2946 elapsed = ext_time - priv->clock_base;
2950 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2955 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2956 RTPJitterBufferItem * item)
2958 guint64 total, elapsed, left, estimated;
2959 GstClockTime out_time;
2960 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2962 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2963 || priv->clock_base == -1 || priv->clock_rate <= 0)
2966 /* compute the elapsed time */
2967 elapsed = compute_elapsed (jitterbuffer, item);
2969 /* do nothing if elapsed time doesn't increment */
2970 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2973 priv->last_elapsed = elapsed;
2975 /* this is the total time we need to play */
2976 total = priv->npt_stop - priv->npt_start;
2977 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2978 GST_TIME_ARGS (total));
2980 /* this is how much time there is left */
2981 if (total > elapsed)
2982 left = total - elapsed;
2986 /* if we have less time left that the size of the buffer, we will not
2987 * be able to keep it filled, disabled buffering then */
2988 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2989 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2990 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2991 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2994 /* this is the current time as running-time */
2995 out_time = item->dts;
2998 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3000 /* if there is almost nothing left,
3001 * we may never advance enough to end up in the above case */
3002 if (total < GST_SECOND)
3003 estimated = GST_SECOND;
3007 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3008 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3010 if (estimated != -1 && priv->estimated_eos != estimated) {
3011 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3012 priv->estimated_eos = estimated;
3016 /* take a buffer from the queue and push it */
3017 static GstFlowReturn
3018 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3020 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3021 GstFlowReturn result = GST_FLOW_OK;
3022 RTPJitterBufferItem *item;
3023 GstBuffer *outbuf = NULL;
3024 GstEvent *outevent = NULL;
3025 GstQuery *outquery = NULL;
3026 GstClockTime dts, pts;
3028 gboolean do_push = TRUE;
3032 /* when we get here we are ready to pop and push the buffer */
3033 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3037 case ITEM_TYPE_BUFFER:
3039 /* we need to make writable to change the flags and timestamps */
3040 outbuf = gst_buffer_make_writable (item->data);
3042 if (G_UNLIKELY (priv->discont)) {
3043 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3044 * into the jitterbuffer so we can modify now. */
3045 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3046 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3047 priv->discont = FALSE;
3049 if (G_UNLIKELY (priv->ts_discont)) {
3050 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3051 priv->ts_discont = FALSE;
3055 gst_segment_position_from_running_time (&priv->segment,
3056 GST_FORMAT_TIME, item->dts);
3058 gst_segment_position_from_running_time (&priv->segment,
3059 GST_FORMAT_TIME, item->pts);
3061 /* apply timestamp with offset to buffer now */
3062 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3063 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3065 /* update the elapsed time when we need to check against the npt stop time. */
3066 update_estimated_eos (jitterbuffer, item);
3068 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3070 case ITEM_TYPE_LOST:
3071 priv->discont = TRUE;
3075 case ITEM_TYPE_EVENT:
3076 outevent = item->data;
3078 case ITEM_TYPE_QUERY:
3079 outquery = item->data;
3083 /* now we are ready to push the buffer. Save the seqnum and release the lock
3084 * so the other end can push stuff in the queue again. */
3086 priv->last_popped_seqnum = seqnum;
3087 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3089 msg = check_buffering_percent (jitterbuffer, percent);
3096 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3099 case ITEM_TYPE_BUFFER:
3101 GST_DEBUG_OBJECT (jitterbuffer,
3102 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3103 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3104 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3105 result = gst_pad_push (priv->srcpad, outbuf);
3107 JBUF_LOCK_CHECK (priv, out_flushing);
3109 case ITEM_TYPE_LOST:
3110 case ITEM_TYPE_EVENT:
3111 /* We got not enough consecutive packets with a huge gap, we can
3112 * as well just drop them here now on EOS */
3113 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3114 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3115 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3116 g_queue_clear (&priv->gap_packets);
3119 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3120 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3123 gst_pad_push_event (priv->srcpad, outevent);
3125 gst_event_unref (outevent);
3127 result = GST_FLOW_OK;
3129 JBUF_LOCK_CHECK (priv, out_flushing);
3131 case ITEM_TYPE_QUERY:
3135 res = gst_pad_peer_query (priv->srcpad, outquery);
3137 JBUF_LOCK_CHECK (priv, out_flushing);
3138 result = GST_FLOW_OK;
3139 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3140 JBUF_SIGNAL_QUERY (priv, res);
3149 return priv->srcresult;
3153 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3155 /* Peek a buffer and compare the seqnum to the expected seqnum.
3156 * If all is fine, the buffer is pushed.
3157 * If something is wrong, we wait for some event
3159 static GstFlowReturn
3160 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3162 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3163 GstFlowReturn result;
3164 RTPJitterBufferItem *item;
3166 guint32 next_seqnum;
3168 /* only push buffers when PLAYING and active and not buffering */
3169 if (priv->blocked || !priv->active ||
3170 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3171 return GST_FLOW_WAIT;
3174 /* peek a buffer, we're just looking at the sequence number.
3175 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3176 * wait for a timeout or something to change.
3177 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3178 item = rtp_jitter_buffer_peek (priv->jbuf);
3183 /* get the seqnum and the next expected seqnum */
3184 seqnum = item->seqnum;
3186 return pop_and_push_next (jitterbuffer, seqnum);
3189 next_seqnum = priv->next_seqnum;
3191 /* get the gap between this and the previous packet. If we don't know the
3192 * previous packet seqnum assume no gap. */
3193 if (G_UNLIKELY (next_seqnum == -1)) {
3194 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3195 /* we don't know what the next_seqnum should be, the chain function should
3196 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3197 * fires, so wait for that */
3198 result = GST_FLOW_WAIT;
3200 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3202 if (G_LIKELY (gap == 0)) {
3203 /* no missing packet, pop and push */
3204 result = pop_and_push_next (jitterbuffer, seqnum);
3205 } else if (G_UNLIKELY (gap < 0)) {
3206 /* if we have a packet that we already pushed or considered dropped, pop it
3207 * off and get the next packet */
3208 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3209 seqnum, next_seqnum);
3210 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3212 result = GST_FLOW_OK;
3214 /* the chain function has scheduled timers to request retransmission or
3215 * when to consider the packet lost, wait for that */
3216 GST_DEBUG_OBJECT (jitterbuffer,
3217 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3218 next_seqnum, seqnum, gap);
3219 result = GST_FLOW_WAIT;
3227 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3229 return GST_FLOW_EOS;
3231 return GST_FLOW_WAIT;
3237 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3239 GstClockTime rtx_retry_timeout;
3240 GstClockTime rtx_min_retry_timeout;
3242 if (priv->rtx_retry_timeout == -1) {
3243 if (priv->avg_rtx_rtt == 0)
3244 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3246 /* we want to ask for a retransmission after we waited for a
3247 * complete RTT and the additional jitter */
3248 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3250 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3252 /* make sure we don't retry too often. On very low latency networks,
3253 * the RTT and jitter can be very low. */
3254 if (priv->rtx_min_retry_timeout == -1) {
3255 rtx_min_retry_timeout = priv->packet_spacing;
3257 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3259 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3261 return rtx_retry_timeout;
3265 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3266 GstClockTime rtx_retry_timeout)
3268 GstClockTime rtx_retry_period;
3270 if (priv->rtx_retry_period == -1) {
3271 /* we retry up to the configured jitterbuffer size but leaving some
3272 * room for the retransmission to arrive in time */
3273 if (rtx_retry_timeout > priv->latency_ns) {
3274 rtx_retry_period = 0;
3276 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3279 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3281 return rtx_retry_period;
3284 /* the timeout for when we expected a packet expired */
3286 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3289 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3291 guint delay, delay_ms, avg_rtx_rtt_ms;
3292 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3293 GstClockTime rtx_retry_period;
3294 GstClockTime rtx_retry_timeout;
3297 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3298 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3300 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3301 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3303 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3304 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3305 GST_TIME_ARGS (rtx_retry_period));
3307 delay = timer->rtx_delay + timer->rtx_retry;
3309 delay_ms = GST_TIME_AS_MSECONDS (delay);
3310 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3311 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3312 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3314 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3315 gst_structure_new ("GstRTPRetransmissionRequest",
3316 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3317 "running-time", G_TYPE_UINT64, timer->rtx_base,
3318 "delay", G_TYPE_UINT, delay_ms,
3319 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3320 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3321 "period", G_TYPE_UINT, rtx_retry_period_ms,
3322 "deadline", G_TYPE_UINT, priv->latency_ms,
3323 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3324 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3326 priv->num_rtx_requests++;
3327 timer->num_rtx_retry++;
3329 GST_OBJECT_LOCK (jitterbuffer);
3330 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3331 timer->rtx_last = gst_clock_get_time (clock);
3332 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3334 timer->rtx_last = now;
3336 GST_OBJECT_UNLOCK (jitterbuffer);
3338 /* calculate the timeout for the next retransmission attempt */
3339 timer->rtx_retry += rtx_retry_timeout;
3340 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3341 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3342 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3343 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3344 if ((priv->rtx_max_retries != -1
3345 && timer->num_rtx_retry >= priv->rtx_max_retries)
3346 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3347 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3348 /* too many retransmission request, we now convert the timer
3349 * to a lost timer, leave the num_rtx_retry as it is for stats */
3350 timer->type = TIMER_TYPE_LOST;
3351 timer->rtx_delay = 0;
3352 timer->rtx_retry = 0;
3354 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3355 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3358 gst_pad_push_event (priv->sinkpad, event);
3364 /* a packet is lost */
3366 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3369 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3370 GstClockTime duration, timestamp;
3371 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3374 RTPJitterBufferItem *item;
3376 seqnum = timer->seqnum;
3377 timestamp = apply_offset (jitterbuffer, timer->timeout);
3378 duration = timer->duration;
3379 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3380 duration = priv->packet_spacing;
3381 lost_packets = MAX (timer->num, 1);
3382 num_rtx_retry = timer->num_rtx_retry;
3384 /* we had a gap and thus we lost some packets. Create an event for this. */
3385 if (lost_packets > 1)
3386 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3387 seqnum + lost_packets - 1);
3389 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3391 priv->num_late += lost_packets;
3392 priv->num_rtx_failed += num_rtx_retry;
3394 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3396 /* we now only accept seqnum bigger than this */
3397 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3398 priv->next_in_seqnum = next_in_seqnum;
3400 /* create paket lost event */
3401 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3402 gst_structure_new ("GstRTPPacketLost",
3403 "seqnum", G_TYPE_UINT, (guint) seqnum,
3404 "timestamp", G_TYPE_UINT64, timestamp,
3405 "duration", G_TYPE_UINT64, duration,
3406 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3408 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3409 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3411 /* remove timer now */
3412 remove_timer (jitterbuffer, timer);
3414 JBUF_SIGNAL_EVENT (priv);
3420 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3423 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3425 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3426 remove_timer (jitterbuffer, timer);
3428 /* there was no EOS in the buffer, put one in there now */
3429 queue_event (jitterbuffer, gst_event_new_eos ());
3431 JBUF_SIGNAL_EVENT (priv);
3437 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3440 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3442 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3444 /* timer seqnum might have been obsoleted by caps seqnum-base,
3445 * only mess with current ongoing seqnum if still unknown */
3446 if (priv->next_seqnum == -1)
3447 priv->next_seqnum = timer->seqnum;
3448 remove_timer (jitterbuffer, timer);
3449 JBUF_SIGNAL_EVENT (priv);
3455 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3458 gboolean removed = FALSE;
3460 switch (timer->type) {
3461 case TIMER_TYPE_EXPECTED:
3462 removed = do_expected_timeout (jitterbuffer, timer, now);
3464 case TIMER_TYPE_LOST:
3465 removed = do_lost_timeout (jitterbuffer, timer, now);
3467 case TIMER_TYPE_DEADLINE:
3468 removed = do_deadline_timeout (jitterbuffer, timer, now);
3470 case TIMER_TYPE_EOS:
3471 removed = do_eos_timeout (jitterbuffer, timer, now);
3477 /* called when we need to wait for the next timeout.
3479 * We loop over the array of recorded timeouts and wait for the earliest one.
3480 * When it timed out, do the logic associated with the timer.
3482 * If there are no timers, we wait on a gcond until something new happens.
3485 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3487 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3488 GstClockTime now = 0;
3491 while (priv->timer_running) {
3492 TimerData *timer = NULL;
3493 GstClockTime timer_timeout = -1;
3496 /* If we have a clock, update "now" now with the very
3497 * latest running time we have. If timers are unscheduled below we
3498 * otherwise wouldn't update now (it's only updated when timers
3499 * expire), and also for the very first loop iteration now would
3500 * otherwise always be 0
3502 GST_OBJECT_LOCK (jitterbuffer);
3503 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3505 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3506 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3508 GST_OBJECT_UNLOCK (jitterbuffer);
3510 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3511 GST_TIME_ARGS (now));
3513 len = priv->timers->len;
3514 for (i = 0; i < len; i++) {
3515 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3516 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3517 gboolean save_best = FALSE;
3519 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3520 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3522 /* find the smallest timeout */
3523 if (timer == NULL) {
3525 } else if (timer_timeout == -1) {
3526 /* we already have an immediate timeout, the new timer must be an
3527 * immediate timer with smaller seqnum to become the best */
3528 if (test_timeout == -1
3529 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3530 timer->seqnum) > 0))
3532 } else if (test_timeout == -1) {
3533 /* first immediate timer */
3535 } else if (test_timeout < timer_timeout) {
3538 } else if (test_timeout == timer_timeout
3539 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3540 timer->seqnum) > 0)) {
3541 /* same timer, smaller seqnum */
3545 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3547 timer_timeout = test_timeout;
3550 if (timer && !priv->blocked) {
3552 GstClockTime sync_time;
3555 GstClockTimeDiff clock_jitter;
3557 if (timer_timeout == -1 || timer_timeout <= now) {
3558 do_timeout (jitterbuffer, timer, now);
3559 /* check here, do_timeout could have released the lock */
3560 if (!priv->timer_running)
3565 GST_OBJECT_LOCK (jitterbuffer);
3566 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3568 GST_OBJECT_UNLOCK (jitterbuffer);
3569 /* let's just push if there is no clock */
3570 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3571 now = timer_timeout;
3575 /* prepare for sync against clock */
3576 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3577 /* add latency of peer to get input time */
3578 sync_time += priv->peer_latency;
3580 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3581 " with sync time %" GST_TIME_FORMAT,
3582 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3584 /* create an entry for the clock */
3585 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3586 priv->timer_timeout = timer_timeout;
3587 priv->timer_seqnum = timer->seqnum;
3588 GST_OBJECT_UNLOCK (jitterbuffer);
3590 /* release the lock so that the other end can push stuff or unlock */
3593 ret = gst_clock_id_wait (id, &clock_jitter);
3596 if (!priv->timer_running) {
3597 gst_clock_id_unref (id);
3598 priv->clock_id = NULL;
3602 if (ret != GST_CLOCK_UNSCHEDULED) {
3603 now = timer_timeout + MAX (clock_jitter, 0);
3604 GST_DEBUG_OBJECT (jitterbuffer,
3605 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3606 GST_STIME_ARGS (clock_jitter));
3608 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3610 /* and free the entry */
3611 gst_clock_id_unref (id);
3612 priv->clock_id = NULL;
3614 /* no timers, wait for activity */
3615 JBUF_WAIT_TIMER (priv);
3620 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3625 * This funcion implements the main pushing loop on the source pad.
3627 * It first tries to push as many buffers as possible. If there is a seqnum
3628 * mismatch, we wait for the next timeouts.
3631 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3633 GstRtpJitterBufferPrivate *priv;
3634 GstFlowReturn result = GST_FLOW_OK;
3636 priv = jitterbuffer->priv;
3638 JBUF_LOCK_CHECK (priv, flushing);
3640 result = handle_next_buffer (jitterbuffer);
3641 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3642 /* now wait for the next event */
3643 JBUF_WAIT_EVENT (priv, flushing);
3644 result = GST_FLOW_OK;
3646 } while (result == GST_FLOW_OK);
3647 /* store result for upstream */
3648 priv->srcresult = result;
3649 /* if we get here we need to pause */
3655 result = priv->srcresult;
3662 JBUF_SIGNAL_QUERY (priv, FALSE);
3665 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3666 gst_flow_get_name (result));
3667 gst_pad_pause_task (priv->srcpad);
3668 if (result == GST_FLOW_EOS) {
3669 event = gst_event_new_eos ();
3670 gst_pad_push_event (priv->srcpad, event);
3676 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3677 * some sanity checks and then emit the handle-sync signal with the parameters.
3678 * This function must be called with the LOCK */
3680 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3682 GstRtpJitterBufferPrivate *priv;
3683 guint64 base_rtptime, base_time;
3685 guint64 last_rtptime;
3687 guint64 ext_rtptime, diff;
3688 gboolean valid = TRUE, keep = FALSE;
3690 priv = jitterbuffer->priv;
3692 /* get the last values from the jitterbuffer */
3693 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3694 &clock_rate, &last_rtptime);
3696 clock_base = priv->clock_base;
3697 ext_rtptime = priv->ext_rtptime;
3699 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3700 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3701 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3702 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3704 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3705 /* we keep this SR packet for later. When we get a valid RTP packet the
3706 * above values will be set and we can try to use the SR packet */
3707 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3710 /* we can't accept anything that happened before we did the last resync */
3711 if (base_rtptime > ext_rtptime) {
3712 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3715 /* the SR RTP timestamp must be something close to what we last observed
3716 * in the jitterbuffer */
3717 if (ext_rtptime > last_rtptime) {
3718 /* check how far ahead it is to our RTP timestamps */
3719 diff = ext_rtptime - last_rtptime;
3720 /* if bigger than 1 second, we drop it */
3721 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3723 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3724 clock_rate, 1000)) {
3725 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3726 /* should drop this, but some RTSP servers end up with bogus
3727 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3728 * so still trigger rptbin sync but invalidate RTCP data
3729 * (sync might use other methods) */
3732 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3733 G_GUINT64_FORMAT, last_rtptime, diff);
3739 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3743 s = gst_structure_new ("application/x-rtp-sync",
3744 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3745 "base-time", G_TYPE_UINT64, base_time,
3746 "clock-rate", G_TYPE_UINT, clock_rate,
3747 "clock-base", G_TYPE_UINT64, clock_base,
3748 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3749 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3751 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3752 gst_buffer_replace (&priv->last_sr, NULL);
3754 g_signal_emit (jitterbuffer,
3755 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3757 gst_structure_free (s);
3759 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3760 gst_buffer_replace (&priv->last_sr, NULL);
3764 static GstFlowReturn
3765 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3768 GstRtpJitterBuffer *jitterbuffer;
3769 GstRtpJitterBufferPrivate *priv;
3770 GstFlowReturn ret = GST_FLOW_OK;
3772 GstRTCPPacket packet;
3773 guint64 ext_rtptime;
3775 GstRTCPBuffer rtcp = { NULL, };
3777 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3779 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3780 goto invalid_buffer;
3782 priv = jitterbuffer->priv;
3784 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3786 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3789 /* first packet must be SR or RR or else the validate would have failed */
3790 switch (gst_rtcp_packet_get_type (&packet)) {
3791 case GST_RTCP_TYPE_SR:
3792 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3798 gst_rtcp_buffer_unmap (&rtcp);
3800 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3803 /* convert the RTP timestamp to our extended timestamp, using the same offset
3804 * we used in the jitterbuffer */
3805 ext_rtptime = priv->jbuf->ext_rtptime;
3806 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3808 priv->ext_rtptime = ext_rtptime;
3809 gst_buffer_replace (&priv->last_sr, buffer);
3811 do_handle_sync (jitterbuffer);
3815 gst_buffer_unref (buffer);
3821 /* this is not fatal but should be filtered earlier */
3822 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3823 ("Received invalid RTCP payload, dropping"));
3829 /* this is not fatal but should be filtered earlier */
3830 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3831 ("Received empty RTCP payload, dropping"));
3832 gst_rtcp_buffer_unmap (&rtcp);
3838 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3839 gst_rtcp_buffer_unmap (&rtcp);
3846 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3849 gboolean res = FALSE;
3850 GstRtpJitterBuffer *jitterbuffer;
3851 GstRtpJitterBufferPrivate *priv;
3853 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3854 priv = jitterbuffer->priv;
3856 switch (GST_QUERY_TYPE (query)) {
3857 case GST_QUERY_CAPS:
3859 GstCaps *filter, *caps;
3861 gst_query_parse_caps (query, &filter);
3862 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3863 gst_query_set_caps_result (query, caps);
3864 gst_caps_unref (caps);
3869 if (GST_QUERY_IS_SERIALIZED (query)) {
3870 RTPJitterBufferItem *item;
3873 JBUF_LOCK_CHECK (priv, out_flushing);
3874 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3875 RTP_JITTER_BUFFER_MODE_BUFFER) {
3876 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3877 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3878 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3880 JBUF_SIGNAL_EVENT (priv);
3881 JBUF_WAIT_QUERY (priv, out_flushing);
3882 res = priv->last_query;
3884 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3889 res = gst_pad_query_default (pad, parent, query);
3897 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3905 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3908 GstRtpJitterBuffer *jitterbuffer;
3909 GstRtpJitterBufferPrivate *priv;
3910 gboolean res = FALSE;
3912 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3913 priv = jitterbuffer->priv;
3915 switch (GST_QUERY_TYPE (query)) {
3916 case GST_QUERY_LATENCY:
3918 /* We need to send the query upstream and add the returned latency to our
3920 GstClockTime min_latency, max_latency;
3922 GstClockTime our_latency;
3924 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3925 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3927 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3928 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3929 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3931 /* store this so that we can safely sync on the peer buffers. */
3933 priv->peer_latency = min_latency;
3934 our_latency = priv->latency_ns;
3937 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3938 GST_TIME_ARGS (our_latency));
3940 /* we add some latency but can buffer an infinite amount of time */
3941 min_latency += our_latency;
3944 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3945 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3946 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3948 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3952 case GST_QUERY_POSITION:
3954 GstClockTime start, last_out;
3957 gst_query_parse_position (query, &fmt, NULL);
3958 if (fmt != GST_FORMAT_TIME) {
3959 res = gst_pad_query_default (pad, parent, query);
3964 start = priv->npt_start;
3965 last_out = priv->last_out_time;
3968 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3969 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3970 GST_TIME_ARGS (last_out));
3972 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3973 /* bring 0-based outgoing time to stream time */
3974 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3977 res = gst_pad_query_default (pad, parent, query);
3981 case GST_QUERY_CAPS:
3983 GstCaps *filter, *caps;
3985 gst_query_parse_caps (query, &filter);
3986 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3987 gst_query_set_caps_result (query, caps);
3988 gst_caps_unref (caps);
3993 res = gst_pad_query_default (pad, parent, query);
4001 gst_rtp_jitter_buffer_set_property (GObject * object,
4002 guint prop_id, const GValue * value, GParamSpec * pspec)
4004 GstRtpJitterBuffer *jitterbuffer;
4005 GstRtpJitterBufferPrivate *priv;
4007 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4008 priv = jitterbuffer->priv;
4013 guint new_latency, old_latency;
4015 new_latency = g_value_get_uint (value);
4018 old_latency = priv->latency_ms;
4019 priv->latency_ms = new_latency;
4020 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4021 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4024 /* post message if latency changed, this will inform the parent pipeline
4025 * that a latency reconfiguration is possible/needed. */
4026 if (new_latency != old_latency) {
4027 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4028 GST_TIME_ARGS (new_latency * GST_MSECOND));
4030 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4031 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4035 case PROP_DROP_ON_LATENCY:
4037 priv->drop_on_latency = g_value_get_boolean (value);
4040 case PROP_TS_OFFSET:
4042 priv->ts_offset = g_value_get_int64 (value);
4043 priv->ts_discont = TRUE;
4048 priv->do_lost = g_value_get_boolean (value);
4053 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4056 case PROP_DO_RETRANSMISSION:
4058 priv->do_retransmission = g_value_get_boolean (value);
4061 case PROP_RTX_NEXT_SEQNUM:
4063 priv->rtx_next_seqnum = g_value_get_boolean (value);
4066 case PROP_RTX_DELAY:
4068 priv->rtx_delay = g_value_get_int (value);
4071 case PROP_RTX_MIN_DELAY:
4073 priv->rtx_min_delay = g_value_get_uint (value);
4076 case PROP_RTX_DELAY_REORDER:
4078 priv->rtx_delay_reorder = g_value_get_int (value);
4081 case PROP_RTX_RETRY_TIMEOUT:
4083 priv->rtx_retry_timeout = g_value_get_int (value);
4086 case PROP_RTX_MIN_RETRY_TIMEOUT:
4088 priv->rtx_min_retry_timeout = g_value_get_int (value);
4091 case PROP_RTX_RETRY_PERIOD:
4093 priv->rtx_retry_period = g_value_get_int (value);
4096 case PROP_RTX_MAX_RETRIES:
4098 priv->rtx_max_retries = g_value_get_int (value);
4101 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4103 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4106 case PROP_MAX_DROPOUT_TIME:
4108 priv->max_dropout_time = g_value_get_uint (value);
4111 case PROP_MAX_MISORDER_TIME:
4113 priv->max_misorder_time = g_value_get_uint (value);
4116 case PROP_RFC7273_SYNC:
4118 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4119 g_value_get_boolean (value));
4123 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4129 gst_rtp_jitter_buffer_get_property (GObject * object,
4130 guint prop_id, GValue * value, GParamSpec * pspec)
4132 GstRtpJitterBuffer *jitterbuffer;
4133 GstRtpJitterBufferPrivate *priv;
4135 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4136 priv = jitterbuffer->priv;
4141 g_value_set_uint (value, priv->latency_ms);
4144 case PROP_DROP_ON_LATENCY:
4146 g_value_set_boolean (value, priv->drop_on_latency);
4149 case PROP_TS_OFFSET:
4151 g_value_set_int64 (value, priv->ts_offset);
4156 g_value_set_boolean (value, priv->do_lost);
4161 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4169 if (priv->srcresult != GST_FLOW_OK)
4172 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4174 g_value_set_int (value, percent);
4178 case PROP_DO_RETRANSMISSION:
4180 g_value_set_boolean (value, priv->do_retransmission);
4183 case PROP_RTX_NEXT_SEQNUM:
4185 g_value_set_boolean (value, priv->rtx_next_seqnum);
4188 case PROP_RTX_DELAY:
4190 g_value_set_int (value, priv->rtx_delay);
4193 case PROP_RTX_MIN_DELAY:
4195 g_value_set_uint (value, priv->rtx_min_delay);
4198 case PROP_RTX_DELAY_REORDER:
4200 g_value_set_int (value, priv->rtx_delay_reorder);
4203 case PROP_RTX_RETRY_TIMEOUT:
4205 g_value_set_int (value, priv->rtx_retry_timeout);
4208 case PROP_RTX_MIN_RETRY_TIMEOUT:
4210 g_value_set_int (value, priv->rtx_min_retry_timeout);
4213 case PROP_RTX_RETRY_PERIOD:
4215 g_value_set_int (value, priv->rtx_retry_period);
4218 case PROP_RTX_MAX_RETRIES:
4220 g_value_set_int (value, priv->rtx_max_retries);
4224 g_value_take_boxed (value,
4225 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4227 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4229 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4232 case PROP_MAX_DROPOUT_TIME:
4234 g_value_set_uint (value, priv->max_dropout_time);
4237 case PROP_MAX_MISORDER_TIME:
4239 g_value_set_uint (value, priv->max_misorder_time);
4242 case PROP_RFC7273_SYNC:
4244 g_value_set_boolean (value,
4245 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4249 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4254 static GstStructure *
4255 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4259 JBUF_LOCK (jbuf->priv);
4260 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4261 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4262 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4263 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4264 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4265 JBUF_UNLOCK (jbuf->priv);